[Serusers] SER + Asterisk - sr-users

25 May 2006

Hi all,

I'm trying to deploy a scenario where I have an openser as a sip proxy
(taking care of registrations) in front of an asterisk (to handle pstn call
routing/billing/etc).

Both openser and asterisk authenticate users on the same database, and I'm
able to register a UA on both.

The problem is when I try to relay calls from openser to asterisk. I always
get the following error:
SIP/2.0 407 Proxy Authentication Required

What am I missing here?

My openser.cfg is the following:

debug=3         # debug level (cmd line: -dddddddddd)
fork=yes
#log_stderror=no # (cmd line: -E)

# Uncomment these lines to enter debugging mode 
#fork=no
log_stderror=yes

listen=192.168.64.102
sip_warning=no

#advertised_address=192.168.1.102

reply_to_via=no
check_via=no    # (cmd. line: -v)
dns=no           # (cmd. line: -r)
rev_dns=no      # (cmd. line: -R)
port=5060
children=4
fifo="/tmp/ser_fifo"

# ------------------ module loading ----------------------------------

# Uncomment this if you want to use SQL database
loadmodule "/usr/lib/openser/modules/mysql.so"

loadmodule "/usr/lib/openser/modules/sl.so"
loadmodule "/usr/lib/openser/modules/tm.so"
loadmodule "/usr/lib/openser/modules/rr.so"
loadmodule "/usr/lib/openser/modules/maxfwd.so"
loadmodule "/usr/lib/openser/modules/usrloc.so"
loadmodule "/usr/lib/openser/modules/registrar.so"
loadmodule "/usr/lib/openser/modules/nathelper.so"
loadmodule "/usr/lib/openser/modules/textops.so"
loadmodule "/usr/lib/openser/modules/xlog.so"
loadmodule "/usr/lib/openser/modules/auth.so"
loadmodule "/usr/lib/openser/modules/auth_db.so"

# -- usrloc params --
# Flush every 60 sec
modparam("usrloc", "db_mode", 0)

# -- auth params --
modparam("auth_db", "db_url", "XXX")
modparam("auth_db", "user_column", "accountcode")
#modparam("auth_db", "domain_column", "accountcode")
modparam("auth_db", "calculate_ha1", 1)
modparam("auth_db", "password_column", "secret")

# -- replication auth param --
modparam("auth", "secret", "mysecret")

# -- rr params --
# add value to ;lr param to make some broken UAs happy
modparam("rr", "enable_full_lr", 1)

route{

        # initial sanity checks -- messages with
        # max_forwards==0, or excessively long requests
        if (!mf_process_maxfwd_header("10")) {
                sl_send_reply("483","Too Many Hops");
                return;
        };
        if ( msg:len > max_len ) {
                sl_send_reply("513", "Message too big");
                return;
        };

        force_rport();
        fix_nated_contact();

        if(uri == myself)
        { 
                if(method == "REGISTER") 
                {
                        # Make sure they are a valid user on our proxy
                        if(!www_authorize("", "astaccount")) 
                        {
                                xlog("L_INFO", "Req Auth For %ct, URI =
%ru\n");
                                www_challenge("", "1");
                                return;
                        };

                        xlog("L_INFO", "Registered Contact %ct,  URI =
%ru\n");
                        save("location");

                        return;
                };
        };

        record_route();  

        # ------------------------------------------------------------------

        # loose-route processing
        if(loose_route())
        {
                xlog("L_INFO", "loose_route(): Looking up %rm URI %ru from
%is\n");
                lookup("location");
                xlog("L_INFO", "loose_route(): t_relay() %rm to URI
%ru\n");
                t_relay();
                return;
        };

        if(method == "INVITE")
            {

                # Assume it came from one of our VoIP phones, all routing is
done by Asterisk
                        xlog("L_INFO", "%rm came from a VoIP phone (%is),
attempting to authorize %fu\n");

                        # Make sure they are a valid user on our proxy
                        if (!www_authorize("", "astaccount"))
                        {
                                   www_challenge("", "0");
                                   xlog("L_INFO", "Failed to authorize %fu
(%is)\n");
                                   return;
                        };

                        # Found a match, this is going to a VoIP phone
                        xlog("L_INFO", "Auth OK, sending URI %ru to
Asterisk
for routing\n");
                        rewritehostport("192.168.64.103:5060");        

                xlog("L_INFO", "INVITE New URI = %ru, t_relay()ing
now\n");
                t_relay();
                # forward(uri:host, uri:port);
                return;
        };

        # ------------------------------------------------------------------

        # forward to current uri now; use stateful forwarding; that
        # works reliably even if we forward from TCP to UDP
        lookup("location");
        xlog("L_INFO", "Default t_relay() (method = %rm, URI = %ru,
>From =
%is)\n");
        if(!t_relay()) 
        {
                xlog("L_INFO", "Failed sending requesting %rm URI
(%ru)\n");
                sl_reply_error();
        };
}

Thanks for your help